[Freeswitch-users] FS - MjSip no voice

[can_man at gmx.de]

Hallo,

thank you for your answer Anthony.

> 
> starting at line 192 you seem to be sending yourself a notify, not sure
> how you did that.

That is indeed strange, I have looked at the MjSip code but haven't found the cause yet.

> you are not by any chance trying to call a registered endpoint using the
> FS
> ip together with @ are you?
> say you fs box is 1.2.3.4 and the phone is registered as 1000
> 
> If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you
> would
> use sofia/internal/1000%1.2.3.4
> The % tells it to resolve the domain as a locally hosted domain and
> translate it to the registered contact instead of using dns.
> 

For testing I at the moment send the incoming call to the voicemail of user 1000 with this code:

return '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\
        '''<document type="freeswitch/xml">\n'''\
        '''<section name="dialplan" description="RE Dial Plan For FreeSwitch">\n'''\
        '''<context name="public">\n'''\
        '''<extension name="voicemail%s">\n'''\
        '''<condition field="destination_number" expression="^(%s)$">\n'''\
        '''<action application="voicemail" data="default $${domain} %s"/>\n'''\
        '''</condition>\n'''\
        '''</extension>\n'''\
        '''</context>\n'''\
        '''</section>\n'''\
        '''</document>''' % (didNumber, didNumber, id)


Works fine with a normal SIP client.
I have captured more output with debug enabled and have also captured the SIP messages originating from MjSip.

FS: http://pastebin.freeswitch.org/8045
MjSip: http://pastebin.freeswitch.org/8046

Thank you very much for your help.
Best wishes,
Phil

> 
> 
> On Sun, Mar 29, 2009 at 5:09 PM, <can_man at gmx.de> wrote:
> 
> > Hello everyone,
> >
> > I am trying to get FS working with the MjSip Java Sip-stack, the
> SipToSis
> > source and the normal one. Everything works well within my own network
> and
> > when using x-lite, but when it comes to making calls from MjSip to an
> > outside FS server I don't hear any voice - seems to be a NAT problem or
> some
> > kind of other MjSip problem. Registration works fine though and SIP
> messages
> > get through ok, but non of the UDP RTP ones. Would be great if someone
> could
> > advice me on how to do the setup correctly.
> >
> > The whole FS trace can be found here:
> http://pastebin.freeswitch.org/8029
> >
> > The settings for MjSip are:
> >
> > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090",
> > "transport_protocols=udp tcp","from_url=<sip:puli at 91.101.58.142:5090>",
> >
> >
> "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",
> >
> >
> "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",
> >
> >
> "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500",
> > "bin_rat=rat","bin_vic=vic"
> >
> >
> > Thank you very much.
> > Best wishes,
> > Phil
> >
> > --
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> >
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> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
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-- 
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